- VOIP EQUIPMENT.
- USB Phone
- VoIP Gateway
- Trunking Gateway
- SIP Proxy Server
- Billing System
- SIP IVR Server
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| Physical Connectors | Cabling | Standard support | Operating Voltage & Operating Temperature |
Operating Humidity & Certification |
| One USB port | USB A-type connector | 1. Conform
to USB 12 Mbps Spec. Version 1.1 2. Conform to USB Audio Device Release 1.0 |
4.5V ~ 5.25V. | 20% ~ 80% RH /CE, FCC |
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| Call GateWay | ||||
| Physical interface | Network and Protocol | Voice feature | Call function | Firmware and configuration update |
| 1.RJ-45 A. WAN X 1 for connecting to HUB or ATU-R directly B. LAN X 1 for PC connection 2.RJ-11 A. Phone X 1 for ATA-171 B. Phone X 2 for ATA-172 C. Phone X 1, Line X 1 for ATA-171P/ATA-171M • Dimension: 9.9 X 9.9 X 3.2 cm |
• SIP v1(RFC2543), v2(RFC 3261)
• IP/ICMP/ARP/RARP/SNTP • TFTP Client/DHCP Client/PPPoE Client • Telnet/HTTP Server • DHCP Server for LAN Port • NAT traversal • Support ToS • Security • HTTP 1.1 basic/digest authentication for Web • setup • MD5 for SIP authentication (RFC2069/RFC2617) |
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Voice codec G.711 : 64k bit/s (PCM) G.726 : 16k/24k/32l/40k bit/s (ADPCM) G.729A : 8k bit/s (CS-ACELP) G.729B : adds VAD & CNG to G.729 • VAD CNG LEC Packet Loss Compensation DTMF In-Band DTMF Out-Band DTMF SIP Info • Tone generation Ring Tone Ring Back Tone Dial Tone Busy Tone Programming Tone |
• Call Hold • Call Waiting • Call Forward • Caller ID • Flash • Volume Adjustment • Speed dial key • Phone book • Call Transfer between FXS, FXO and IP port (ATA- 171M only) • Call Forwarding between FXS, FXO and IP port (ATA- 171M only) |
Web Browser Telnet Voice configuration TFTP HTTP |
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| H.323/SIP Version | ||||
| Protocol | Compatible Server | Telephony Features | FAX support | Audio Codec support |
| ITU-T H.323 v2/v3/v4 compliance | GnuGK(Behind NAT) | Call Hold, Call Transfer, Call Forward (H.450) | Automatically FAX detection
T.38 G3 FAX Group 3 FAX relay at 2.4 - 14.4 kbps |
G.711A/μ-law, G.723.1, G.729A, G.729, G.729B, G.729AB |
| Network Interface | FXS Interface | Voice Quality | QoS | Caller ID |
| 10/100Base-T Ethernet RJ-45 port x 2 | •
RJ-11 Telephone port (FXS) x 2 for WellGate 3502A • RJ-11 Telephone port (FXS) x 4 for WellGate 3504A only • 2-wire loop start • Programmable AC impedance, Feeding voltage, Ring Voltage, Ring Cadence, Loop current and call progress tone • ON-Hook Voltage: 48Vdc • Ring Voltage: 50 V RMS • Loop Current: Constant 23mA (Support Payphone Charge signal 12K/16K/Polarity Reversal+) |
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VAD (Voice Activity Detection) • CNG (Comfort Noise Generation) • AEC (Acoustic Echo Cancellation) -- G.168 • Dynamic Jitter Buffer |
DiffServ | FSK (Bellcore)/DTMF generation |
| Console Port | Tone | Network Support | Security | Configuration |
| 1 D-SUB 9 pin RS-232 port | DTMF / CPT (Call Progress Tone) generation/detection | • Statically and DHCP for IP address assignment • Behind NAT Router or IP sharing device • PPPoE | Command line interface and Web management interface Password protected | Console port, TELNET and Web Browser |
| System Upgrade | Power | Certification | Operation Temp | Humidity |
| Firmware upgrade through network by TFTP/FTP | Input AC 100V~240V Output DC12V | FCC Part 15 Class B, VCCI Class FCC Part 15 Class B, CE Class B, VCCI Class B | 0° C to 40° C | 10% to 90% (Non-condensing) |
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| 5260-T1/E1 Trunking Gateway | ||||
| Introduction | Key Features | |||
| WellGate 5260 is an UNIVERSAL VOIP GATEWAY which navigates the calls in between H.323, SIP and PSTN freely , not simple PSTN to VOIP calls or vice verse. It can easily implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP and SIP, H.323 and H.323 calls simultane- ously. With built-in PSTN/VOIP IVR helps service provider to establish their own voice service platform quick. It is suitable for application of Pre-paid Calling card service termination. The built-in both ways IVR provide flexible function for termination and Pin code authentication announcement. | WellGate 5260 provides
three models as follows. • WellGate 5260-1, one line T1/E1 trunk gateway • WellGate 5260-2, two lines T1/E1 trunk gateway • WellGate 5260-4, four lines T1/E1 trunk gateway |
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Navigate Call Freely in SIP, H.323 and PSTN • Support SIP RFC 3261 and ITU-T H.323 V5 Simultaneously • Up to 4 Programmable E1/T1 Trunks • PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG , SS7 • Support Audio Codec G.711, G.723.1, G.729A, GSM • SIP to PSTN Call and vice versa • H.323 to PSTN Call and vice versa • H.323 to SIP Call and vice versa • SIP to SIP Call • H.323 to H.323 Call • Support up-to 16 Multiple SIP Proxy Servers |
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Support SIP Proxy, Gatekeeper and P2P Calls Simultaneously • Built-in Universal VOIP Address Book • Support Early Media and SIP Delay Media • Support RADIUS Authentication, Authorization and Accounting • Intelligent PSTN Call Routing and in-Trunk Hunting • Flexible Digit Manipulation Plan • Support Calling/Called Number Replacement • In-band and out of Band DTMF Transmission • T.38 Fax Relay up to 14400 bps • Dynamic Call Treatment Based on Drag and Drop Call Flow Editor |
•
Built-in PSTN and VOIP IVR • Provides Call Detail Record • Full Web Management Interface • Comply with RoHS • Support up-to 16 SIP Register/Outbound Proxy Servers • Support Flexible VOIP Routing and Account Code • Support IP Access Control List with Account Code • Support IP, ANI, DNIS ACL Group • Added VOIP Hunting Features • Support VOIP Hunting based on Priority & Weekday/Time • Support PSTN Line Hunting based on Time • Made in Taiwan |
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SIP 6500 - Enterprise Certification Power/Environmental Processors & Storage Front Panel Display LAN Interface CE FCC Power 90~240V auto switch Operating temperature 0~60°C Relative humidity 5%~95% Intel Pentium 4 Windows XP Embedded 1024 MB RAM 512 MB DOM Upgradeable OS Upgradeable AP Image Power LED Dom Access LED System Ready LED LCD Status 10/100/1000 Base Ethernet x 2 SIP service & management IP address Standard Protocol
SupportSIP Registrar SIP Outbound Proxy
ServerSubscriber Management System Capacity
(Standard version)• RFC 3261
• RFC 2976
• RFC 3262
• RFC 2327
• RFC 2833
• RFC 3581
• RFC 3264
• RFC 3265
• draft-ietf-sip-cc-transfer-05
• Support SIP UDP
• Support SIP URI
• sProgrammable SIP Service Port• Dynamic Register
• Predefine User (up to 2 URI)
• Predefine NAT User
• MD5 Registrar Authentication
• Register Type Selection
• Register to External Softswitch or Proxy (multiple)
• Effective / Expired Date
• Global or Subscriber based Register TTL
• Programmable Nonce Live Time
• Programmable Authentication Check Period
• Programmable General Max Register Time
• Programmable Max NAT Register Time• Stateful Proxy Server
• Support Call-based Authentication MD5/Radius
• Sequential Call Forking
• Parallel Call Forking (Programmable ring or answer)
• Proxy Peering Support
• Hierarchical Proxy Support
• Subscriber / System based Response Timer (no answer, first response)
• Auto NAT User Detect• Subscriber Access Control
• Separate Web Password
• User Group
• Subscriber logon
• Disallow NAT register• Max Subscribers Support: 500
• Max Concurrent Call: 100
• Max Concurrent RTP Support for NAT user: 100NAT Traversal Network Management System Capacity(Mini version) User Manual
Physical Dimensions• NAT Traversal Support for Outbound and Inbound
• Automatically NAT detection and RTP Proxy
• External RTP Proxy Resource Support**
• Preferred NAT Resource Server**
• Support enhanced NAT Partition
• Intelligent RTP Proxy Resource Management
• sProgrammable SIP Service Port• DHCP
• Fixed IP
• DNS, Dynamic DNS
• Ping
• SNMP V2 MIB I & II
• SNMP get command, set command
• SNMP Trap SupportMax Subscribers Support: 200
Max Concurrent Call: 50
Max Concurrent RTP Support for NAT user: 50English User Guide
483mm(W) x 450mm(D) x 44mm(H) Weight: 6.92Kg** contact welltech for availability
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6600-Billing Server Key Features RADIUS AAA Support
1. Authentication Message .
2. Authorization Message.
3. Billing Start/Stop Message
• RFC 2865, 2866 Compliance with Selected Attributes .
• Up-to 500k subscribers support.
• Fully Web Management Interface.
• Support Prepaid/Postpaid User.
• Max Call Duration Protection .
• Immediate Response.
• Share Secret with MD5 Protection.
• Automatic 6500 provision.
• Detail Access Log .Up-to 5 level User Management .
• Administrator .
• Distributor .
• Group Reseller .
• Reseller .
• Subscriber .
Prepaid Service • Real Time Balance Deduct .
• Subscriber/Reseller Recharge & Rollback .
• Recharge Log .
• Effective Date/Expired Date .
• PIN Code Generate and Consume .Postpaid Service
• Call Detail Record Storage
• Effective Date/Expired Date
• CDR Report External Database support
• MSSQL
• Built-in DB Connection Pool management
• Provide Basic Report .
• Support Coin Phone Service .
• Support Calling ID (ANI) Validation .
• Support Charge Account .
• Auto Monthly Charge Deduction .Flexible Rate Plan Support
• Up-to 5 Charge Segments per Rate Prefix
• Effective Date/Expired Date
• Longest Prefix Match
• Programmable charge unit, amount and cycle
• Support Per Call Charge
• Call Screening
• Holiday & Night Time Charge
• Free Monthly Minutes based on Prefix
• Monthly Free Charge based on Prefix
• Deductible Monthly FeeWeb Browser HTTP
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SIP IVR Server 6800 Key Features • Up-to 120 Universal VOIP Channels
• SIP RFC 3261 Compliance
• Fully Web Management Interface
• Audio Codec G.711, G.729A, G.723.1*
• Drag and Drop Call Flow Editor
• Real Time Status/Variable debugger
URich-set of predefined components:
• Basic flow components
• IVR components
• Database components
• Flow control components
• RADIUS components
• Channel components
• HTTP Access Components
• External Customized components• Support Call Hold and Transfer
• Support in-Band and out-of-Band DTMF relay
• Support Database Connection Pools
• Support Internal/External Job Push & Retrieve
• Support Internal/External Hook Function Calls
• Free Text Math Expression with rich functions
• Hitless Call Flow Update
• Optimized Developing Platform
*G.723.1 is an optional for 6800S
| 1. Maintaining an unwavering commitment to customer and partner friendly business practices. | Technical Support | ||
| 2. Remaining passionate about VoIP and IP communications technology. | If you believe your product is DOA (Dead on Arrival) please contact | ||
| 3. Delivering valuable expertise through consultation and education. | E - Mail: support@quickcomglobal.com |
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| 4. Being accessible and welcoming of a phone call. | In case of emergency, please give us a call at 86.411.3970.7285. | ||
| 5. Developing channels for clear two-way communications. | We are available from 9am - 6pm EST
Monday through Saturday. |
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| 6. Creating and delivering on VoIP solutions that evolve to meet your changing needs. | More info contact by MSN&E-mail:eko@quickcomglobal.com |